It is well known that digital recording of audio signals, such as music signals, affords numerous advantages during playback. One of the major advantages is the absence, during sound reproduction, of the intrinsic noise of the physical medium on which the audio information is recorded. Since digital recording stores the audio information in the form of numbers, it is possible to recover the same information during playback, hence in a way independent of the playback means. Of course, there exists an intrinsic noise of the digital signal, such as quantization distortion, but it is possible to prove that this noise is in fact inaudible once certain conditions at the level of digital recording are observed. Instead, analog recording requires the audio information to be “written” in the form of an analog signal. In most cases the reproduction medium is a magnetic tape, which is polarized with a magnetic field determined by the analog signal. In this way, it should be possible to read the same signal during playback by reading the polarization of the tape. Unfortunately a magnetic tape, even if it is of the best quality, in any case presents in regard to an analog playback system a residual level of polarization which can be perceived in the form of a low-level wideband noise also referred to as tape noise.
Nevertheless, analog audio recording is today widely used for example in the car-radio and cassette-recorder sector. Even though it is possible to record a digital signal on a magnetic tape, there exists a large number of car radios and cassette recorders which are in fact incompatible with, and hence not suited for, reproduction of audio material in a digital format. In addition, there is an enormous amount of audio cassettes recorded in analog format which thus require the availability of analog playback equipment.
In the past various noise-reduction systems have been proposed aimed at reducing the effects of tape noise. Such systems usually convert the analog signal into a new one, which is less sensitive to tape noise. This conversion is referred to as “encoding”. During playback, the encoded signal is not designed to be reproduced as it is, in so far as the encoding process, which is designed to render the signal less sensitive to tape noise, modifies (sometimes to a significant extent) the audio information. For this reason, the encoded signal is normally subjected to a second conversion, which is carried out at the playback stage, referred to as “decoding”. The decoding operation is designed to reproduce in a dual way, i.e., in a complementary way, the encoding process and to reconstruct the original recorded signal as faithfully as possible.
Clearly, proper decoding can be performed only if the encoding system enables a decoder to be made that is able to reconstruct the original audio signal without error, i.e., exactly counterbalancing the action of the encoder. Otherwise, decoding becomes inherently prone to reconstruction errors. A class of noise-reduction systems that has met with a great deal of success is based on the idea of boosting the audio signal during the encoding process, in particular when the dynamic level is low. In this way, the encoded signal is less sensitive to low-level additional noise, such as precisely tape noise, than is the original audio signal. Decoding implies an action of reduction of the level of the encoded signal, in such a way that the original audio information is restored. Using this technique, any other additional signal superimposed on the encoded signal, such as tape noise, is attenuated.
Since tape noise is predominant at higher frequencies, as in case of compact cassettes, the encoding/decoding process described above can be performed only where these frequency components are present, leaving the other signal components unaltered. This approach lies at the basis of noise-reduction techniques known as Dolby techniques, and in particular the Dolby-B technique. These are techniques widely used for noise reduction, as is described, for example, in the publication “Dolby Licensee Information Manual: Noise Reduction,” issue 6, Dolby Laboratories Licensing Corporation (1999).
Since the music industry continues to produce compact cassettes recorded in analog format, noise-reduction decoders of the same type as Dolby-B decoders implemented, for example, in the car-radio industry, have so far been built exclusively using analog-signal processing techniques (i.e., using circuits comprising resistors, capacitors and other analog electronic components).
Over the last few years, the car-radio industry has identified in digital signal processors (DSPs) the most convenient alternative to dedicated analog circuits. In fact, any currently available analog circuit designed specifically for a given purpose may be reproduced most effectively in the form of an algorithm which is run on a DSP purposely programmed for emulating the given circuit. In this way, most of the functions available in a car radio, in particular reduction of the tape noise, can be performed by a DSP, provided that an appropriate software capable of emulating the functions of the analog circuit is made available. This software can be updated, redesigned, reconfigured and modified according to software evolution, industry and market trends, in a way altogether independent of the processing device itself. Furthermore, since digital-signal processing device are constantly decreasing in cost and increasing in performance, it is possible also for the device to be updated, leaving the functions, configurations and performance of the software unaltered. In addition, digital-signal processing devices are largely insensitive to variations in manufacturing lots and variations due to temperature given that they basically carry out arithmetic operations instead of implementing transfer functions of an analog type. This fact constitutes a further advantage as regards implementation based on digital processors, since in noise-reduction systems the tolerances in regard to variations in component parameters must be very stringent in order to avoid any possible mismatches between encoder and decoder. Such mismatches may lead to perceptible differences between the original audio information and the audio information present in the decoded signal. A digital implementation of the decoder is thus free from any possible mismatches caused by tolerance faults in the decoder components.
What has been said so far is, however, true only up to a point.
Techniques are in fact known which enable design of digital-signal processing systems which merely implement pre-existing analog-signal processing systems. There are moreover known techniques which enable implementation in digital form of analog feedback-controlled networks, where the non-linear components are intrinsically grouped together in one or more control sub-networks. In spite of this, certain noise-reduction systems of the Dolby-B type cannot be implemented simply by converting their blocks into the digital domain, or by grouping together their non-linear blocks into one or more sub-networks designed to control one or more linear blocks. Indeed, the currently accepted idea in the sector is that the operation of decoding encoded signals using noise-reduction systems of a known type, such as Dolby B, cannot be implemented satisfactorily in digital form. This fact is believed to be due to the decoding strategy adopted in systems of the Dolby-B type and similar systems.
FIGS. 1a and 1b of the attached drawings are high-level representations of the structure used by Dolby B and similar systems to encode (FIG. 1a) and decode (FIG. 1b) audio signals. In detail, the audio signal XE 110 to be encoded passes through an encoding block 120, the output of which is added to the original signal by means of an adder 130 so as to produce the encoded signal YE 140. During playback, the signal YD 150 (which is made up of the encoded signal YE 140 and the tape noise) is decoded to produce an audio signal XD 180, in which the tape noise is reduced. The decoding process uses a decoding block 160 identical to the block 120. The signal 180 is fed back to the block 160 so as to produce an encoded version of the reconstructed signal 170, which is to be subtracted from the signal 150 to produce the decoded audio signal.
It can be shown that the decoder in fact reconstructs at output the original audio signal XE 110 from the signal 150—in the absence of tape noise. Intuitively, if the decoder produces a signal 180 identical to the signal 110, then the latter signal can be used to calculate the signal 170. Given the equality between the blocks 120 and 160, the signal 170 can be subtracted from the signal 150 so as to reconstruct the original audio signal. In practice, a noise-reduction system of the same type as the Dolby-B system comes close to satisfying the hypothesis of absence of tape noise, given that it cancels out the typical tape noise of a compact cassette, as this noise may be perceived during playback. Furthermore, this architecture presents a clear practical advantage in terms of implementation of the noise-reduction system by means of analog circuits, since both the encoder and the decoder use the same electronic components.
To illustrate the concept more clearly, in FIGS. 2a and 2b the blocks 120 and 160 have been expanded and represented as conceptually divided into a filtering part and a control part. Both parts of the decoder are identical to homologous parts of the encoder. In the encoder shown in FIG. 2a the filtering part 215 generates two outputs, 220 and 222. The former of these outputs feeds the adder 235 to produce the encoded signal 240. The latter output feeds the control part 225 so as to generate a signal 230, which is used to drive the filtering part. Symmetrically, in the decoder shown in FIG. 2b the two outputs, 265 and 267, of the filtering part 260 are used, respectively, to produce the decoded signal 255 (after subtraction from the signal 250) and to drive the control part 270. As in the previous case, the output 275 of the control part drives the filtering part.
In FIGS. 3a and 3b the high-level structures of the encoder and decoder, respectively, are further expanded. Both the encoder and the decoder require a preliminary stage—blocks 315 and 372, respectively—for eliminating the undesired components of the signal, as well as the other high-frequency signals outside the band of the tape recorder. The filtering part is divided into a linear filter and a non-linear characteristic, the latter being also referred to as overshoot characteristic, the role of which will emerge more clearly from what follows. The signal 330, which corresponds to the signal 222 of FIG. 2a, is produced by the linear filter 325. The signal 352, which corresponds to the signal 220 of FIG. 2a, is produced by processing the signal 330 with the overshoot characteristic 350. Symmetrically, in the decoder shown in FIG. 3b the signal 380, which corresponds to the signal 267 of FIG. 2b, is produced by the linear filter 379. The signal 392, which corresponds to the signal 265 of FIG. 2a, is produced by processing the signal 380 with the overshoot characteristic 390 identical to the overshoot characteristic 350. It should be noted that the subtraction 268 is here performed by the inverter 377. In this way, the side effect is obtained of inverting the sign of the decoded signal 378, without this having any consequences for the audio signal reproduced during playback.
The control part is likewise divided into a control filter and a rectifying stage, also known as non-linear integrator. In the encoder shown in FIG. 3a, the signal 330, which is identical to the signal 222, is processed by a linear control filter 335, the output 340 of which feeds the non-linear integrator 342. This block produces a slowly varying signal 345, which drives the linear filter contained in the filtering part of the encoder. Similarly, in the decoder shown in FIG. 3b, the signal 380, which is identical to the signal 267, is processed by a linear control filter 382, which is identical to the linear control filter 335, and the output 384 of which feeds the non-linear integrator 386, which in turn is identical to the non-linear integrator 342. Also this block produces a slowly varying signal 388, which is identical to the signal 345 and which drives the linear filter contained in the filtering part of the decoder.
The network shown in FIG. 4 represents the non-linear integrator. The voltage-controlled input signal 410 is rectified by two non-linear resistances in parallel, each one of these being obtained by means of a resistor—R415 and R420—and a germanium diode—Ge425 and Ge430, respectively. The rectified signal is then processed by two integration stages: one is a linear resistor/capacitor stage in parallel made up of a capacitor C435 and a resistor R440; the other is a non-linear resistor/capacitor stage in series made up of a capacitor C455 and a non-linear resistance defined by a resistor R450 in parallel with a silicon diode Si445. The control signal 460, which is equal to the level of charge of the capacitor C455, is the signal presented at output.
The non-linear integrator illustrated in FIG. 4 is able to provide a control signal that meets the following requirements:                The rectification speed depends upon the variation in the dynamics of the input signal during the attack phase; i.e., the wider the increase in dynamics on the input, the faster the rectification.        There occurs a linear decay during a release phase; i.e., the control signal drops exponentially in the presence of a decrease in dynamics of the input signal.        There are no discontinuities in the output signal; i.e., the signal 460 presents a smooth evolution even in the presence fast attacks due to large onsets in the input signal.        
Even though the non-linear integrator reacts promptly to large onsets present on the input, it cannot drive the filtering part of the decoder in the presence of large but short onsets of the signal to be encoded. In this case, the non-linear characteristic 350 provides a transient overshoot compression which has the effect of limiting the high—and undesired—dynamics of the signal 330 during onsets that are large but short. Hence, the signal 352 encodes correctly the audio signal also in the presence of large but short onsets of the signal. The role of the overshoot compression 390 at the decoding stage is, as usual, reversed, so that the signal 370 is correctly decoded even during large but short onsets of the signal.